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Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics最新文献

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Regularized estimation of cepstrum envelope from discrete frequency points 离散频率点倒谱包络的正则化估计
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482993
Olivier Cappé, Jean Laroche, Eric Moulines
{"title":"Regularized estimation of cepstrum envelope from discrete frequency points","authors":"Olivier Cappé, Jean Laroche, Eric Moulines","doi":"10.1109/ASPAA.1995.482993","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482993","url":null,"abstract":"This paper presents an improved method for the estimation of a continuous frequency-envelope when the value of this envelope is specified only at discrete frequencies. It is based on the Galas/Rodet (1990) approach which consists of fitting a cepstral amplitude envelope to the specified frequency points by minimizing a frequency-domain least-squares criterion. This paper introduces a regularization technique which increases the robustness of the estimation procedure. Used in combination with a warped frequency-scale, the proposed method is shown to provide an efficient model for the frequency envelope of speech signals.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"45 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114433678","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 80
Two microphone nonlinear frequency domain beamformer for hearing aid noise reduction 用于助听器降噪的双传声器非线性频域波束形成器
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482905
E. Lindemann
{"title":"Two microphone nonlinear frequency domain beamformer for hearing aid noise reduction","authors":"E. Lindemann","doi":"10.1109/ASPAA.1995.482905","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482905","url":null,"abstract":"A two microphone beamforming algorithm for hearing aid noise reduction is described. The algorithm is based on a short time Fourier transform analysis/synthesis system with nonlinear frequency domain adaptive filtering. The algorithm has been implemented on a portable battery powered device. Subjective ratings and BKB sentence scores for a group of hearing aid wearers are reported.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"130 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114943365","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 17
Multi-channel blind signal separation by decorrelation 基于去相关的多路盲信号分离
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482980
Dcb Chan, P. Rayner, S. Godsill
{"title":"Multi-channel blind signal separation by decorrelation","authors":"Dcb Chan, P. Rayner, S. Godsill","doi":"10.1109/ASPAA.1995.482980","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482980","url":null,"abstract":"The separation of independent sources from mixed observed data is a fundamental and challenging problem. In many practical situations, observations may be modelled as linear mixtures of a number of source signals, i.e. a linear multi-input multi-output system. A typical example is speech recordings made in an acoustic environment in the presence of background noise and/or competing speakers. Other examples include EEG signals, passive sonar applications and crosstalk in data communications. We propose iterative algorithms to solve the n/spl times/n linear time invariant system under two different constraints. Some existing solutions for 2/spl times/2 systems are reviewed and compared.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117097124","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 16
Sub-band coding of audio using recursively indexed quantization 音频的子带编码使用递归索引量化
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482987
Yang-Jeng Chen, R. Maher
{"title":"Sub-band coding of audio using recursively indexed quantization","authors":"Yang-Jeng Chen, R. Maher","doi":"10.1109/ASPAA.1995.482987","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482987","url":null,"abstract":"A low-complexity audio data compression technique using sub-band coding (SBC) and a recursively indexed quantizer (RIQ) is proposed. The characteristics of the proposed system are investigated and several relevant implementation issues are described. The objective performance of the proposed system is compared to conventional coding techniques. A real time implementation of the proposed system running on a DSP microprocessor is also discussed. The results indicate that the RIQ-SBC system exhibits objective signal-to-quantization noise ratio 2 to 5 dB higher than other coders of similar computational complexity when processing a variety of wideband audio signals.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"57 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132091712","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 3
Microphone array calibration for robust adaptive processing 麦克风阵列校准鲁棒自适应处理
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482901
M. Hoffman
{"title":"Microphone array calibration for robust adaptive processing","authors":"M. Hoffman","doi":"10.1109/ASPAA.1995.482901","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482901","url":null,"abstract":"Significant interest exists in the use of microphone arrays for enhancing audio and speech signals. Various combinations of fixed array, adaptive array, time-domain and frequency domain processing approaches are being considered. Any well-designed sensor array processing system needs (either explicitly or implicitly) accurate modeling or measurement of the sensor responses to allow robust and effective signal enhancement. The article explores the limits that microphone array calibration places on robust speech enhancement. Modelled microphone array data is used to examine the effect of variations in array parameters on processing algorithms.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131168323","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 1
Limitations of handsfree acoustic echo cancellers due to nonlinear loudspeaker distortion and enclosure vibration effects 由于非线性扬声器畸变和外壳振动效应，免提声回声消除器的局限性
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482968
A. N. Birkett, R. Goubran
{"title":"Limitations of handsfree acoustic echo cancellers due to nonlinear loudspeaker distortion and enclosure vibration effects","authors":"A. N. Birkett, R. Goubran","doi":"10.1109/ASPAA.1995.482968","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482968","url":null,"abstract":"The limitations of adaptive echo cancellers (AEC) for handsfree telephony include noise, finite precision and truncation effects, undermodelling of the acoustic impulse response, vibration of the plastic enclosure, loudspeaker nonlinearities, dynamic tracking and convergence and double-talk. This paper examines the effect that the loudspeaker nonlinearity and enclosure vibration have on the steady state ERLE performance and concludes that enclosure vibration is an important limitation which, although rarely mentioned in the literature, is probably the major limitation in terms of sound quality. Experimental measurements indicate that enclosure vibration at medium to high loudspeaker volumes limits the achievable ERLE and the perceived audio quality more than any other limitation mentioned above.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"34 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131449121","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 78
Estimating azimuth and elevation from interaural differences 根据内部差异估计方位和高度
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482967
K. Martin
{"title":"Estimating azimuth and elevation from interaural differences","authors":"K. Martin","doi":"10.1109/ASPAA.1995.482967","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482967","url":null,"abstract":"Modeling of human auditory localization has largely been limited to lateralization, or left-to-right position. This paper describes an attempt to tackle the more complicated problem of position estimation with two degrees of freedom (azimuth and elevation). Differences in interaural intensity and arrival time are extracted from the acoustic signals at the left and right eardrums, and an estimate of position is formed which is optimal for certain classes of source signals. Examples of \"spatial likelihood maps\" generated by the model are given and the types of errors made by the model are quantified. It is suggested that such a model may work well in conjunction with a spectral cue model like the one suggested by Zakarauskas and Cynader (J. Acoust. Soc. Amer., Vol. 94, 1993, pp. 1323-31).","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"154 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131757628","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 33
Instantaneous and frequency-warped techniques for source separation and signal parametrization 用于源分离和信号参数化的瞬时和频率扭曲技术
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482910
A.L. Wang
{"title":"Instantaneous and frequency-warped techniques for source separation and signal parametrization","authors":"A.L. Wang","doi":"10.1109/ASPAA.1995.482910","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482910","url":null,"abstract":"This paper summarizes several contributions to the study of audio signal parameterization and source separation. The philosophy of frequency-warped signal processing provides powerful means for separating the AM and FM contributions to the MS bandwidth of a complex-valued, frequency-varying sinusoid p[n], transforming it into a signal with slowly-varying parameters. The use of frequency- and harmonic-locked loops to track the phase variation of analytic signals provides the necessary accurate instantaneous frequency information to drive frequency-warped filtering.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132838274","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 9
Some considerations for the design of auditory displays 听觉显示器设计的一些考虑
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482950
R. Gilkey
{"title":"Some considerations for the design of auditory displays","authors":"R. Gilkey","doi":"10.1109/ASPAA.1995.482950","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482950","url":null,"abstract":"Two experiments evaluate the effect of noise on the accuracy of sound localization judgments, with an eye toward the environments where 3-dimensional auditory displays are likely to be implemented. The ability to determine whether a sound came from the front or from the back and whether a sound came from above or from below is disrupted at much higher signal-to-noise ratios than the ability to determine whether a sound came from the left or from the right. Changes in the location of the masker can have complex effects on the pattern of localization errors. A third experiment examined the ability to localize speech signals, as might often be required in auditory displays. The results also indicate that the ability to distinguish front from back and up from down is easily disrupted (i.e., it is worse for speech than for nonspeech signals), but that the ability to determine left from right is robust (i.e., it is similar for speech and for nonspeech).","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"40 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122005449","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 4
Auditory segregation of concurrent signals: an operational definition 并发信号的听觉分离:一个操作定义
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482908
P. Divenyi
{"title":"Auditory segregation of concurrent signals: an operational definition","authors":"P. Divenyi","doi":"10.1109/ASPAA.1995.482908","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482908","url":null,"abstract":"Auditory segregation of simultaneous, multiple (quasi-) steady-state acoustic signals which compose an ensemble implies simultaneous resolution along three \"cardinal dimensions\": vectors of pitch (residue plus timbre), temporal structure, and location. It is proposed that the normalized distance between two, or several, component signals in an ensemble along any of the three dimensions is traded off by differences on the two other dimensions. A psychophysical paradigm is described for objective measurement of auditory segregation and a metric is suggested for the prediction of segregation in the 3D parameter space.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116431487","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 3
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